Convert AU to DSS — Free Online Tool
Convert Sun AU audio files to DSS (Digital Speech Standard) format using ADPCM IMA OKI compression — the native codec used by Olympus, Philips, and Grundig digital dictation devices. This tool is ideal for transcribing or archiving PCM audio from Unix systems into a compact, speech-optimized format compatible with professional dictation workflows.
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FFmpeg Command
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
AU files typically store uncompressed PCM audio (in this case, 16-bit big-endian signed PCM by default), which is a lossless, high-fidelity representation of the audio signal. During conversion to DSS, FFmpeg decodes the PCM stream from the AU container and re-encodes it using the ADPCM IMA OKI codec — a lossy adaptive delta pulse-code modulation algorithm optimized for speech frequencies. The DSS container is purpose-built for dictation, so the output is significantly smaller than the original AU file, but with reduced fidelity that is designed and acceptable for voice recordings rather than music or broadband audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all decoding, codec conversion, and remuxing. In the browser version of this tool, FFmpeg runs via WebAssembly (FFmpeg.wasm) entirely on your local machine — no data is sent to a server. |
-i input.au
|
Specifies the input file in Sun AU format. FFmpeg reads the AU header to determine the audio codec (most commonly pcm_s16be) and decodes the raw PCM audio stream for re-encoding. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the audio stream using the ADPCM IMA OKI codec, which is the only codec supported by the DSS container. This lossy codec applies adaptive delta pulse-code modulation tuned for speech compression, producing a much smaller file than the original PCM audio. |
output.dss
|
Sets the output filename and format. The .dss extension tells FFmpeg to use the Digital Speech Standard muxer, producing a file compatible with Olympus, Philips, and Grundig dictation devices and their associated transcription software. |
Common Use Cases
- Archiving Unix system voice recordings or audio memos into a format compatible with professional digital dictation software like Olympus DSS Player or Philips SpeechExec.
- Preparing speech recordings captured on legacy Sun workstations for ingestion into a legal or medical transcription pipeline that accepts DSS files.
- Reducing the file size of large AU audio logs from Unix telephony or IVR systems before transferring them to dictation-based review workflows.
- Converting historical Unix-era audio announcements or voice notes into the compact DSS format for long-term speech archive storage.
- Migrating AU recordings from a Unix server to a Grundig or Olympus handheld dictation device that natively reads DSS files.
- Batch-converting a folder of AU audio memos to DSS so they can be processed by speech recognition software that accepts DSS input.
Frequently Asked Questions
Yes, because DSS uses the lossy ADPCM IMA OKI codec, which is purpose-built for compressing speech rather than preserving full audio fidelity. If your AU file contains music or broadband audio, the quality loss will be noticeable. For voice recordings — the intended use case of DSS — the degradation is generally acceptable and the format produces very compact file sizes.
The DSS container is a proprietary format jointly developed by Olympus, Philips, and Grundig specifically for digital dictation devices, and it mandates the ADPCM IMA OKI codec as its sole audio codec. Unlike general-purpose containers such as MKV or MP4, DSS has no provision for alternate codecs, so FFmpeg always uses adpcm_ima_oki for any DSS output regardless of the source format.
AU files can contain PCM variants including pcm_s16be, pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. FFmpeg will automatically decode whichever of these codecs is present in your AU file before re-encoding to ADPCM IMA OKI, so the source codec does not require any manual adjustment on your part. However, if your AU file uses a codec that FFmpeg cannot identify or that is corrupted, the conversion may fail.
ADPCM IMA OKI compresses audio at a much lower bitrate than uncompressed PCM, so DSS files are typically a fraction of the size of the source AU file. A 16-bit PCM AU file at 8 kHz uses about 128 kbps, while DSS output is generally in the range of 13–16 kbps, yielding roughly an 8–10x reduction in file size. Actual ratios depend on the sample rate and bit depth of the source AU file.
Yes. On Linux or macOS, you can use a shell loop such as: for f in *.au; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.au}.dss"; done. On Windows Command Prompt, a for loop with similar logic achieves the same result. This is especially useful if you have a folder of legacy Unix audio memos to migrate into a dictation archive.
No. The AU format supports minimal metadata (primarily a text annotation field in the file header), and the DSS format is not designed to carry general-purpose metadata tags. FFmpeg does not map AU header annotations to DSS metadata fields during this conversion, so any embedded text information in the source AU file will be lost in the output.
Technical Notes
The AU format's simplicity — a short header followed by raw PCM data — makes it straightforward for FFmpeg to decode, but the DSS output format introduces significant constraints. ADPCM IMA OKI is a 4-bit ADPCM variant tuned to the narrow frequency range of human speech, and it works best with mono audio at low sample rates (typically 8 kHz), which is standard for dictation devices. If your AU file is stereo or uses a high sample rate, FFmpeg may need to downsample and convert to mono as part of the encoding process to fit the DSS container's expectations. There is no adjustable quality parameter for DSS output (the -b:a flag is not applicable to adpcm_ima_oki in this context), so the codec operates at its fixed compression ratio. DSS files produced this way should be compatible with Olympus DSS Player and Philips SpeechExec, though some proprietary DSS variants (DSS Pro) may have additional header requirements not covered by FFmpeg's muxer.